[asterisk-users] Nat issue one way audio on IP dial
Nasir Javaid
nasirjavaidnasir at gmail.com
Thu Jul 29 03:34:31 CDT 2010
thanks Jim
I will check stun server settings asap,
but i have noticed 192.168.x.x is also present in the debug of successful
call having both way audio. so i don't think this has to do anything with
this.
below is the sip debug of successful call .
---
Audio is at 79.80.154.99 port 14034
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 116.18.35.235:28614:
INVITE sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport
From: "pepsi coke" <sip:12345678901 at 79.80.154.99:5678>;tag=as12245807
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
Contact: <sip:12345678901 at 79.80.154.99:5678>
Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Jul 2010 15:06:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 9626 9626 IN IP4 79.80.154.99
s=session
c=IN IP4 79.80.154.99
t=0 0
m=audio 14034 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Jul 21 11:06:24] WARNING[23749]: chan_sip.c:2872 sip_call: Setting
auto-congest time to 15000 ms.
-- Called adf
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678
Contact: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350
From: "pepsi coke"<sip:12345678901 at 79.80.154.99:5678>;tag=as12245807
Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/adf-00794e30 is ringing
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678
Contact: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350
From: "pepsi coke"<sip:12345678901 at 79.80.154.99:5678>;tag=as12245807
Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 185
v=0
o=- 2 2 IN IP4 192.168.0.12
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.12
t=0 0
m=audio 15956 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.12:15956
Found description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.12:15956
list_route: hop: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
[Jul 21 11:06:38] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99
set_destination: Parsing
<sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
for address/port to send to
set_destination: set destination to 116.18.35.235, port 28614
Transmitting (NAT) to 116.18.35.235:28614:
ACK sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK00fdcc7c;rport
From: "pepsi coke" <sip:12345678901 at 79.80.154.99:5678>;tag=as12245807
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350
Contact: <sip:12345678901 at 79.80.154.99:5678>
Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- Call on SIP/adf-00794e30 left from hold
-- SIP/adf-00794e30 answered SIP/pepsi-9fd06cc0
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
<------------->
--- (0 headers 1 lines) ---
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
SUBSCRIBE sip:adf at ast-server.axvoice.com:5678 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:28614
;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:adf at 116.18.35.235:28614>
To: "adf"<sip:adf at ast-server.axvoice.com:5678>
From: "adf"<sip:adf at ast-server.axvoice.com:5678>;tag=5d297f22
Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
User-Agent: X-Lite release 1104o stamp 56125
Event: message-summary
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 116.18.35.235 : 28614 (NAT)
Found peer 'adf'
Looking for adf in uscan_int (domain ast-server.axvoice.com)
ast-server*CLI>
<--- Transmitting (NAT) to 116.18.35.235:28614 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.12:28614
;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;received=116.18.35.235;rport=28614
From: "adf"<sip:adf at ast-server.axvoice.com:5678>;tag=5d297f22
To: "adf"<sip:adf at ast-server.axvoice.com:5678>;tag=as724c598c
Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Really destroying SIP dialog 'MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.'
Method: SUBSCRIBE
Reliably Transmitting (NAT) to 116.18.35.235:28614:
OPTIONS sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport
From: "asterisk" <sip:asterisk at 79.80.154.99:5678>;tag=as223ef4a7
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
Contact: <sip:asterisk at 79.80.154.99:5678>
Call-ID: 5c66fbdf4234deca50d5c44a18641582 at 79.80.154.99
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Jul 2010 15:07:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport=5678
Contact: <sip:192.168.0.12:28614>
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=15133f38
From: "asterisk"<sip:asterisk at 79.80.154.99:5678>;tag=as223ef4a7
Call-ID: 5c66fbdf4234deca50d5c44a18641582 at 79.80.154.99
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '5c66fbdf4234deca50d5c44a18641582 at 79.80.154.99'
Method: OPTIONS
ast-server*CLI>
<------------>
Scheduling destruction of SIP dialog '
6514fece69f1718e5cefe72632909c0e at 79.80.154.99' in 23936 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 116.18.35.235:28614:
NOTIFY sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport
From: "asterisk" <sip:asterisk at 79.80.154.99:5678>;tag=as756cae64
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
Contact: <sip:asterisk at 79.80.154.99:5678>
Call-ID: 6514fece69f1718e5cefe72632909c0e at 79.80.154.99
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92
Messages-Waiting: no
Message-Account: sip:asterisk at 79.80.154.99 <sip%3Aasterisk at 79.80.154.99>
Voice-Message: 0/0 (0/0)
---
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport=5678
Contact: <sip:192.168.0.12:28614>
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=b9541904
From: "asterisk"<sip:asterisk at 79.80.154.99:5678>;tag=as756cae64
Call-ID: 6514fece69f1718e5cefe72632909c0e at 79.80.154.99
CSeq: 102 NOTIFY
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6514fece69f1718e5cefe72632909c0e at 79.80.154.99'
Method: NOTIFY
[Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:3074 update_call_counter: Call to
peer 'adf' removed from call limit 2
Scheduling destruction of SIP dialog '
25a6e3604896da0e5482a7565560ce3b at 79.80.154.99' in 18624 ms (Method: INVITE)
[Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99
set_destination: Parsing
<sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
for address/port to send to
set_destination: set destination to 116.18.35.235, port 28614
Reliably Transmitting (NAT) to 116.18.35.235:28614:
BYE sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0
Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK05cc42e6;rport
From: "pepsi coke" <sip:12345678901 at 79.80.154.99:5678>;tag=as12245807
To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350
Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
>Date: Wed, 28 Jul 2010 09:36:51 -0700
>From: Jim Dickenson <dickenson at cfmc.com>
>Subject: Re: [asterisk-users] Nat issue one way audio on IP dial
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID: <353D789C-0987-49E1-988E-F3C98E41F8A7 at cfmc.com>
>Content-Type: text/plain; charset="us-ascii"
>Do you have your softphone setup to use a stun server so it can send it's
public IP address in the SIP packets? I see in the SIP >debug output a
192.168 address for the RTP packets to go to which of course will not work.
>--
>Jim Dickenson
>mailto:dickenson at cfmc.com <dickenson at cfmc.com>
>CfMC
>http://www.cfmc.com/
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